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Voir la version complète : asterisk: configuration pour pré décroché automatique.



Hub49
01/11/2016, 17h16
Bonjour à tous et toutes,

Je suis sur une distribution Mageia 5 avec un bureau KDE.

Je viens d'installer via les dépôts asterisk (11.23.1) et les softphones ekiga et jitsi. Les installations se sont bien passées je pense puisque les softphones se lancent correctement: je me vois connecté avec ma ligne sip.
Le but d'une telle installation est de me fournir un standard pour recevoir des appels, lesquels auront un pré décroché d'attente.

En console, j'ai cette info lorsque je tape asterisk -r et je reload ensuite :


localhost*CLI> reload
[Nov 1 15:08:30] WARNING[711]: cel.c:361 do_reload: Could not load cel.conf
[Nov 1 15:08:30] NOTICE[711]: app_queue.c:7953 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action.
[Nov 1 15:08:30] NOTICE[711]: pbx_ael.c:164 pbx_load_module: Starting AEL load process.
[Nov 1 15:08:30] NOTICE[711]: pbx_ael.c:177 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'.
[Nov 1 15:08:30] NOTICE[711]: pbx_ael.c:180 pbx_load_module: AEL load process: checked config file name '/etc/asterisk/extensions.ael'.
[Nov 1 15:08:30] NOTICE[711]: pbx_ael.c:187 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'.
[Nov 1 15:08:30] NOTICE[711]: pbx_ael.c:192 pbx_load_module: AEL load process: merged config file name '/etc/asterisk/extensions.ael'.
[Nov 1 15:08:30] NOTICE[711]: pbx_ael.c:195 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'.
localhost*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
[root@localhost mand]#


En console, j'ai après avoir taper "sip show users" :

localhost*CLI> sip show user
Usage: sip show user <name> [load]
Shows all details on one SIP user and the current status.
Option "load" forces lookup of peer in realtime storage.
localhost*CLI> sip show users
Username Secret Accountcode Def.Context ACL Forcerport
101 azerty local No Yes
102 azerty local No Yes
6000 1234 work No No
localhost*CLI>


Comme j'ai essentiellement bidouillé dans les fichiers extensions.conf et sip.conf, je ne comprends pas le problème évoquant le fichier extensions.ael
Je mets ici les deux fichiers :

extensions.conf (un peu long, désolé):


[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no

[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=DAHDI/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=DAHDI/G2 ; Trunk interface

TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)

;[context]
;exten => someexten,{priority|label{+|-}offset}[(alias)],application(arg1,arg2,...)
; <time range>,<days of week>,<days of month>,<months>[,<timezone>]
;include => daytime,9:00-17:00,mon-fri,*,*
;include => weekend,*,sat-sun,*,*
;include => weeknights,17:02-8:58,mon-fri,*,*
;ignorepat => 9

[dundi-e164-canonical]
;exten => 12564286000,1,Gosub(6000,stdexten(IAX2/foo))
;exten => 12564286000,n,Goto(default,s,1) ; exited Voicemail
;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})

[dundi-e164-customers]
;exten => _12564286000,1,Dial(SIP/customer1)
;exten => _12564286001,1,Dial(IAX2/customer2)

[dundi-e164-via-pstn]
;exten => _1256428XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all of 256-428
;exten => _1256325XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Ditto for 256-325

[dundi-e164-local]
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn

[dundi-e164-switch]

switch => DUNDi/e164

[dundi-e164-lookup]

include => dundi-e164-local
include => dundi-e164-switch

[macro-dundi-e164]

exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup

[iaxtel700]
exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)

[iaxprovider]
;switch => IAX2/user:[key]@myserver/mycontext

[trunkint]
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${FILTER(0-9,${EXTEN:${GLOBAL(TRUNKMSD)}})})

[trunkld]
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXXXXXX,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

[trunklocal]
exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

[trunktollfree]
exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

[international]
ignorepat => 9
include => longdistance
include => trunkint

[longdistance]
ignorepat => 9
include => local
include => trunkld

[local]
ignorepat => 9
include => default
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
include => parkedcalls
; switch => IAX2/user:password@bigserver/local
; lswitch => Loopback/12${EXTEN}@othercontext
; eswitch => IAX2/context@${CURSERVER}
; This is the dialing hook. use:
; include => outbound-freenum

[outbound-freenum]
exten => _X*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)

[outbound-freenum2]
exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)}) ; make sure the suffix is all digits as well
same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" != "${SUFFIX}"]?fn-CONGESTION,1)
; filter out bad characters per the README-SERIOUSLY.best-practices.txt document
same => n,Set(TIMEOUT(absolute)=10800)
same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freen um.org)}) ; perform our lookup with freenum.org
same => n,GotoIf($["${isnresult}" != ""]?from)
same => n,Set(DIALSTATUS=CONGESTION)
same => n,Goto(fn-CONGESTION,1)
same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial) ; check if we set the FREENUMDOMAIN global variable in [global]
same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ; if we did set it, then we'll use it for our outbound dialing domain
same => n(dial),Dial(SIP/${isnresult},40)
same => n,Goto(fn-${DIALSTATUS},1)

exten => fn-BUSY,1,Busy()

exten => _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
same => n,Congestion()

[macro-trunkdial]
; ${ARG1} - What to dial
exten => s,1,Dial(${ARG1})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1,Hangup
exten => _s-.,1,NoOp

[stdexten]
; ${EXTEN} - Extension
; ${ARG1} - Device(s) to ring
; ${ARG2} - Optional context in Voicemail

-------->suite du fichier extensions.conf dans le post suivant car trop long.


et sip.conf :




[general]
context=local ;Contexte par defaut
bindport=5060 ;UDP standard
bindaddr=0.0.0.0 ;bind access to all
srvlookup=yes ;activer les lookup DNS des appels
language=fr ;MSG vocaux en FR

[101] ;Login SIP
secret=azerty ;Mot de passe
callerid= »Bob_les_ponges» <101> ;Affichage lors de l appel
context=local ;appels geres dans extension local
mailbox=101@default ;compte de msg vocale cfr voicemail.conf
type=friend ;allow in et out
host=dynamic ;adresse ip du client
nat=yes ;utiliser derriere du NAT

[102]
secret=azerty
callerid= »Kiki » <102>
context=local
type=friend
host=dynamic
nat=yes
mailbox=102@default


Je suis connecté avec une Bbox. J'ai une ligne sip. Ekiga. J'utilise un casque avec micro pour les appels.
En test, si je m'appelle, ça sonne côté appelant et j'ai le fichier de pré décroché qui se lance côté réception d'appel, soit exactement l'inverse de ce que je souhaite !

Je pense que le problème n'est pas gravissisme mais vu mon niveau général.. je tourne en rond. Il y a forcément un truc que je n'ai pas indiqué...et même plusieurs.

Merci aux intervenants qui me fileront un coup de main pour comprendre le problème.

Hub49
01/11/2016, 17h20
Suite du fichier extensions.conf :


exten => _X.,50000(stdexten),NoOp(Start stdexten)
exten => _X.,n,Set(LOCAL(ext)=${EXTEN})
exten => _X.,n,Set(LOCAL(dev)=${ARG1})
exten => _X.,n,Set(LOCAL(cntx)=${ARG2})
exten => _X.,n,Set(LOCAL(mbx)=${ext}${IF($[!${ISNULL(${cntx})}]?@${cntx})})
exten => _X.,n,Dial(${dev},20) ; Ring the interface, 20 seconds maximum
exten => _X.,n,Goto(stdexten-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => stdexten-NOANSWER,1,Voicemail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start
exten => stdexten-BUSY,1,Voicemail(${mbx},b) ; If busy, send to voicemail w/ busy announce
exten => stdexten-BUSY,n,Return() ; If they press #, return to start
exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1) ; Treat anything else as no answer
exten => a,1,VoicemailMain(${mbx}) ; If they press *, send the user into VoicemailMain
exten => a,n,Return()

[stdPrivacyexten]
; Standard extension subroutine:
; ${ARG1} - Extension
; ${ARG2} - Device(s) to ring
; ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1 extension-priority)
; ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1 extension-priority)`
; ${ARG5} - Context in voicemail (if empty, then "default")
exten => _X.,60000(stdPrivacyexten),NoOp(Start stdPrivacyexten)
exten => _X.,n,Set(LOCAL(ext)=${ARG1})
exten => _X.,n,Set(LOCAL(dev)=${ARG2})
exten => _X.,n,Set(LOCAL(dontcntx)=${ARG3})
exten => _X.,n,Set(LOCAL(tortcntx)=${ARG4})
exten => _X.,n,Set(LOCAL(cntx)=${ARG5})

exten => _X.,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""])
exten => _X.,n,Dial(${dev},20,p) ; Ring the interface, 20 seconds maximum, call screening
; option (or use P for databased call _X.creening)
exten => _X.,n,Goto(stdexten-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => stdexten-NOANSWER,1,Voicemail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
exten => stdexten-NOANSWER,n,NoOp(Finish stdPrivacyexten NOANSWER)
exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start
exten => stdexten-BUSY,1,Voicemail(${mbx},b) ; If busy, send to voicemail w/ busy announce
exten => stdexten-BUSY,n,NoOp(Finish stdPrivacyexten BUSY)
exten => stdexten-BUSY,n,Return() ; If they press #, return to start
exten => stdexten-DONTCALL,1,Goto(${dontcntx},s,1) ; Callee chose to send this call to a polite "Don't call again" script.
exten => stdexten-TORTURE,1,Goto(${tortcntx},s,1) ; Callee chose to send this call to a telemarketer torture script.
exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1) ; Treat anything else as no answer
exten => a,1,VoicemailMain(${mbx}) ; If they press *, send the user into VoicemailMain
exten => a,n,Return

[macro-page];
; ${ARG1} - Device to page
exten => s,1,ChanIsAvail(${ARG1},s) ; s is for ANY call
exten => s,n,GoToIf($[${AVAILSTATUS} = "1"]?autoanswer:fail)
exten => s,n(autoanswer),Set(_ALERT_INFO="RA") ; This is for the PolyComs
exten => s,n,SIPAddHeader(Call-Info: Answer-After=0) ; This is for the Grandstream, Snoms, and Others
exten => s,n,NoOp() ; Add others here and Post on the Wiki!!!!
exten => s,n,Dial(${ARG1})
exten => s,n(fail),Hangup

[demo]
include => stdexten
exten => s,1,Wait(1) ; Wait a second, just for fun
exten => s,n,Answer ; Answer the line
exten => s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
exten => s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds
exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory message
exten => s,n(instruct),BackGround(demo-instruct) ; Play some instructions
exten => s,n,WaitExten ; Wait for an extension to be dialed.
exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
exten => 2,n,Goto(s,instruct)
exten => 3,1,Set(CHANNEL(language)=fr) ; Set language to french
exten => 3,n,Goto(s,restart) ; Start with the congratulations
exten => 1000,1,Goto(default,s,1)
exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
; (but skip if channel is not up)
exten => 1234,n,Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)}) )
exten => 1234,n,Goto(default,s,1) ; exited Voicemail
exten => 1235,1,Voicemail(1234,u) ; Right to voicemail
exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,n,Voicemail(1234,b) ; Unless busy
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
exten => #,n,Hangup ; Hang them up.
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default) ; Call the Asterisk demo
exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo site
exten => 500,n,Goto(s,6) ; Return to the start over message.
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,n,Echo ; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it's over
exten => 600,n,Goto(s,6) ; Start over
exten => 76245,1,Macro(page,SIP/Grandstream1)
exten => _7XXX,1,Macro(page,SIP/${EXTEN})
exten => 7999,1,Set(TIMEOUT(absolute)=60)
exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n,d)
exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
;exten => 1265,1,Dial(Phone/phone0,15)
;exten => 1265,n,Goto(s,5)

[page]
exten => _X.,1,Macro(page,SIP/${EXTEN})
;[mainmenu]
;exten => s,1,Answer
;exten => s,n,Background(thanks) ; "Thanks for calling press 1 for sales, 2 for support, ..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(submenu,s,1)
;exten => 2,1,Hangup
;include => default
;[submenu]
;exten => s,1,Ringing ; Make them comfortable with 2 seconds of ringback
;exten => s,n,Wait,2
;exten => s,n,Background(submenuopts) ; "Thanks for calling the sales department. Press 1 for steve, 2 for..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(default,steve,1)
;exten => 2,1,Goto(default,mark,2)

[public]
; ATTENTION: If your Asterisk is connected to the internet and you do
; not have allowguest=no in sip.conf, everybody out there may use your
; public context without authentication. In that case you want to
; double check which services you offer to the world.
;
include => demo
[default]
[local]
exten => _1XX, 1, Dial(SIP/${EXTEN}, 15) ; Compose 101 appelle franky etc
exten => _1XX, n, VoiceMail(${EXTEN}) ; Voicemail apres 15 secondes
exten => 90,1,VoiceMailMain(${CALLERID(num)}) ; Messagerie
exten => 300, 1, Meetme(300)

include => demo
;exten => _41X.,1,Dial(SIP/${FILTER(0-9,${EXTEN:2})}@sipprovider,,r)
;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1(Joe Schmoe) ; Channel hints for presence
;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer
;exten => 6245,n(dial),Dial(${HINT},20,rtT) ; Use hint as listed
;exten => 6245,n,Voicemail(6245,u) ; Voicemail (unavailable)
;exten => 6245,s+1,Hangup ; s+1, same as n
;exten => 6245,dial+101,Voicemail(6245,b) ; Voicemail (busy)
;exten => 6361,1,Dial(IAX2/JaneDoe,,rm) ; ring without time limit
;exten => 6389,1,Dial(MGCP/aaln/1@192.168.0.14)
;exten => 6390,1,Dial(JINGLE/caller/callee) ; Dial via jingle using labels
;exten => 6391,1,Dial(JINGLE/asterisk@digium.com/mogorman@astjab.org) ;Dial via jingle using asterisk as the transport and calling mogorman.
;exten => 6394,1,Dial(Local/6275/n) ; this will dial ${MARK}

;exten => 6275,1,Gosub(${EXTEN},stdexten(${MARK}))
; assuming ${MARK} is something like DAHDI/2
;exten => 6275,n,Goto(default,s,1) ; exited Voicemail
;exten => mark,1,Goto(6275,1) ; alias mark to 6275
;exten => 6536,1,Gosub(${EXTEN},stdexten(${WIL}))
; Ditto for wil
;exten => 6536,n,Goto(default,s,1) ; exited Voicemail
;exten => wil,1,Goto(6236,1)
;exten => 6600,hint,park:701@parkedcalls
;exten => 6600,1,noop
; You can also monitor the status of a queue by providing a hint for a
; particular queue name.
;exten => 8502,hint,Queue:markq
;exten => 8502,1,Queue(markq)

;To subscribe to the availability of a free member in the 'markq' queue.
;Note: '_avail' is added to the QueueName
;exten => 8501,hint,Queue:markq_avail
;exten => 8501,1,Queue(markq)
; Some other handy things are an extension for checking voicemail via
; voicemailmain
;exten => 8500,1,VoicemailMain
;exten => 8500,n,Hangup
;exten => 8600,1,Meetme(1234)
;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
;[acme-incoming]
;exten => s,1,Wait(1)
;exten => s,n,Answer()
;exten => s,n(menu),Playback(acme/vm-brief-menu)
;exten => s,n(exten),Background(vm-enter-num-to-call)
;exten => s,n,WaitExten(5)
;exten => s,n(goodbye),Playback(vm-goodbye)
;exten => s,n(end),Hangup()
;include => acme-extens
;exten => i,1,Playback(vm-invalid)
;exten => i,n,Goto(s,exten) ; optionally, transfer to operator
;exten => t,1,Goto(s,goodbye)
; this is the context our internal SIP hardphones use (see sip.conf)
;[acme-internal]
;exten => s,1,Answer()
;exten => s,n(exten),Background(vm-enter-num-to-call)
;exten => s,n,WaitExten(5)
;exten => s,n(goodbye),Playback(vm-goodbye)
;exten => s,n(end),Hangup()
;include => trunkint
;include => trunkld
;include => trunklocal
;include => acme-extens
;exten => 777,1,DISA(no-password,acme-incoming)
;[acme-extens]
;include => stdexten
;exten => 111,1,Gosub(111,stdexten(SIP/pete_1,acme))
;exten => 111,n,Goto(s,exten)
;exten => 112,1,Gosub(112,stdexten(SIP/nancy_1,acme))
;exten => 112,n,Goto(s,end)


Pour le prédécroché apparemment :

pour le prédécroché c'est l'option "m" de la commande "dial": exten => 1,1,Dial(SIP/toto,,m(musique))

jean
02/11/2016, 21h31
- le fichier extensions.ael a été installé avec les paquets, dans l'absolu, tu peux soit le vider, soit l'effacer - ca contient du code qui si utilisé, apporte des fonctionnalités d'un pabx

- attention, softphone + asterisk sur le meme serveur = conflit pour utiliser le port 5060.... il faut t'assurer que le sofptohne n'essaie pas d'utiliser le port 5060 comme port source